There are a number of existing techniques for time-scale/pitch modification of audio signals which are known in the art. These can be broadly classified as follows.
(a) Time domain methods:
These techniques attempt to estimate the fundamental period of a musical signal by detecting periodic activity in the audio signal. By this process, an input signal is delayed and multiplied by the undelayed signal, the product of which is then smoothed in a low pass filter to provide an approximate measure of the auto-correlation function. The autocorrelation function is then used to detect a nonperiodic signal or a weak periodic signal which might be hidden in the noise. Once the fundamental period of the musical signal is found the process is repeated and the analyzed sections of the signal are overlapped. A significant disadvantage in these techniques is that most audio signals do not have a fundamental period. For example polyphonic instruments, recordings with reverberation and percussion sounds do not have an identifiable fundamental period. Further, when applying such methods, transients in the music are repeated. This leads to notes having multiple starts and ends. Another problem with this technique is that overlapping of the delayed sections of the music can produce an audio effect which is metallic, mechanical or exhibits echo-like nature.
(b) Sinusoidal analysis methods:
These techniques assume that the input signal is made up from pure sinusoids. The inherent disadvantage of such a method is therefore self evident.
Sinusoidal analysis techniques use Short Time Fast Fourier Transforms (FFT) to estimate the frequency of the component sinusoids. The derived signal is then synthesized with a bank of tone generators to produce the desired output. Short Time Fourier Analysis captures information about the frequency content of a signal within a time interval, governed by the Window Function chosen. A significant disadvantage of such techniques is that a single time-domain window is applied to all the frequency content of the signal, so the signal analysis cannot correspond accurately to human perception of the signal content. Also, conventional sinusoidal analysis methods use a local maxima search of the magnitude spectrum to determine the frequency of the constituent sinusoids including consideration of relative phase changes between analysis frames. This technique ignores any side-band information located around each of the local maxima. The effect of this is to exclude any signal modulation occurring within a single analysis frame, resulting in a smearing of the sound and almost a complete loss of transients. An example of such a transient, in the audio context, is a guitar pluck.
(c) Phase vocoder methods:
This type of technique uses a Fast Fourier Transform as a large bank of filters and treats the output of each of the filters separately. The relative phase change between two consecutive analyses of the input are used to estimate the frequency of the signal content within each bin. A resulting frequency-domain signal is synthesized from this information, treating each bin as a separate signal. In contrast to sinusoidal analysis techniques, this method retains the spectral energy distribution of the original signal. However, it destroys the relative phase of any transient information. Therefore, the resulting sound is smeared and echo-like.
In view of the prior art techniques, it would therefore be desirable to analyze and process audio signals so that the resultant output retains the tonal characteristics of the original signal and is capable of accurately capturing transient sounds without smearing or introducing an echo-like character to the output signal.
Accordingly, it is an object of the present invention to provide a technique for processing audio signals which achieves the abovementioned aims and ameliorates at least some of the disadvantages inherent in the prior art or at least provides the public with a useful choice. Further, it is an object of the invention to provide a signal analysis and synthesis method which can also be applied to the coding of signals in general.